2 Session Initiation Protocol - An application layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony. - IETF RFC3261 User location: Determination of the end system to be used for communication User availability: Determination of the willingness of called party to communicate User capabilities: Determination of the media and parameters to be used Session setup: ringing, establishment of session parameters at both called and calling party Session management: including transfer and termination of sessions, modifying session parameters, and invoking services 5 6 Based on HTTP-like request/response transaction model SIP typically runs on UDP for performance Client request invokes function on server At least one response Uses most HTTP header fields, encoding rules, and status codes Own reliability mechanisms May also use TCP May use Transport Layer Security (TLS) protocol for secure connection Readable format for displaying information Uses concepts similar to recursive and iterative searches of DNS Incorporates Session Description Protocol (SDP) Defines session content using types similar to MIME 7 8

3 SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture. SIP Components RTP transports real time data and provides QoS feedback RTSP controls delivery of streaming media. MEGACO controls gateways to the Public Switched Telephone Network (PSTN). SDP describes multimedia sessions Location Redirect Registrar PSTN User Agent Proxy Proxy Gateway 9 10 Clients send SIP requests (initiates a call) and receive responses. s receive SIP requests and send responses Both servers and clients can terminate calls Acts as both a server and a client. Responds to requests directly or passes them on to other servers. Interprets, rewrites or translates a request before forwarding it

4 Provides information about a called party's possible location(s) Used when a user cannot be found at his/her normal address. Returns zero or more new addresses to the client. Does not initiate its own SIP requests. Does not accept or terminate calls Accepts register requests and uses the received information to update data at a location server. May support authentication Typically co-located with a proxy or redirect server and may offer location services. 15 SIP components communicate by exchanging SIP messages: SIP Methods: INVITE Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER Registers the user agent. OPTIONS Used to query the capabilities of a server. INFO Used to carry out-ofbound information, such as DTMF digits. SIP Responses: 1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Failure Responses. 6xx - Global Failures Responses. 16

5 SIP borrows much of the syntax and semantics from HTTP. A SIP messages looks like an HTTP message message formatting, header and MIME support. An example SIP header: SIP Header INVITE SIP/2.0 Via: SIP/2.0/UDP :5060 From: To: Call-ID: CSeq: 100 INVITE Expires: 180 User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled Accept: application/sdp Contact: Content-Type: application/sdp The SIP address is identified by a SIP URI, in the format: Examples of SIP URI:s: Establishing communication using SIP usually occurs in six steps: 1. Registering, initiating and locating the user. 2. Determine the media to use involves delivering a description of the session that the user is invited to. 3. Determine the willingness of the called party to communicate the called party must send a response message to indicate willingness to communicate accept or reject. 4. Call setup. 5. Call modification or handling example, call transfer (optional). 6. Call termination. Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server. Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location. The registration information is periodically refreshed and each user client must re-register with the proxy/registration server. Typically the proxy/registration server will forward this information to be saved in the location/redirect server. SIP Phone User REGISTER 200 Proxy/ Registration REGISTER 200 SIP Messages: REGISTER Registers the address listed in the To header field. 200 OK. Location/ Redirect 19 20

7 Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications. H.323 defines: Call establishment and teardown. Audio visual or multimedia conferencing. ITU-T Recommendation H.323 Version H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service. Gatekeeper Multipoint Control Unit Terminal Packet Based Networks Gateway Circuit Switched Networks 27 28

10 Call agent or media gateway controller Provides call signaling, control and processing intelligence to the gateway. Sends and receives commands to/from the gateway. Gateway Provides translations between circuit switched networks and packet switched networks. Sends notification to the call agent about endpoint events. Execute commands from the call agents. Call Agent or Media Gateway Controller (MGC) MGCP Media Gateway (MG) SIP H.323 Call Agent or Media Gateway Controller (MGC) MGCP Media Gateway (MG) MGCP: A master/slave protocol. Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent). Used between call agents and media gateways. Differs from SIP and H.323 which are peer-topeer protocols. Interoperates with SIP and H A protocol that is evolving from MGCP and developed jointly by ITU and IETF: Megaco - IETF. H.248 or H.GCP - ITU. SIP and H.323 are comparable protocols that provide call setup, call teardown, call control, capabilities exchange, and supplementary features. MGCP is a protocol for controlling media gateways from call agents. In a VoIP system, MGCP can be used with SIP or H.323. SIP or H.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways

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